FIG. 4 shows a typical layout of a prior art network of this type. In this Figure, symbols 10a, 10b, and 10c are ATM exchanges, symbols 30a, 30b, and 30c are exchanges equipped with a voice compression function (PBXs), and symbols 50-1, 50-2, 50-3, 50-4, 50-5, and 50-6 are telephone sets. Also, the dotted line portions in ATM exchanges 10a, 10b, 10c and PBXs 30a, 30b, 30c are through-routes; in particular, symbol 35 in PBX 35 indicates a relay route when a voice signal is relay-exchanged by the corresponding PBX 30b.
Next, the case will be described in which compressed voice is transferred by such a prior art network. For example, when a call is made between telephone set 50-1 and telephone set 50-3, a telephone call originated by telephone set 50-1 is transferred in the order: PBX 30a, ATM exchange 10a, ATM exchange lob, and PBX 30b, before arriving at telephone set 50-3. In this case, all the processing of the call connection information is performed between PBX 30a and PBX 30b.
The operation of PBX 30a, ATM exchange 10a, ATM exchange 10b, and PBX 30b in this case will now be described in more detail. In the present description, it will be assumed that connection between PBX 30a and ATM exchange 10a and connection between ATM exchange 10b and PBX 30b is effected by a typical interface TTC 2 Mbps interface.
FIG. 5 shows an example of the frame format of a TTC 2 Mbps interface; this is constituted by providing a frame synchronization bit F at the head, followed by one 64 kbps channel, a call control ch (channel), voice ch (channels) 1 to 30, and, in addition, an undefined channel [see FIG. 2(a)] 1ch, not particularly shown in this Figure.
In the case where voice compression is not performed by PBX 30A, PCM (64 kbps) voice is accommodated by the above voice ch, so these channels are of 8-bit type (8 Khz.times.8 bit).).
In contrast, if voice compression is performed by PBX 30A (in this example, 16 kbps compression), compressed voice allocation (16 kbps compressed voice) as shown for example in FIG. 6 is performed within each voice ch (channel); compressed voice is entered in the amount of 2 bits at the head, invalid data being inserted for the remaining 6 bits.
Usually, when a voice signal is transferred by PBX 30A without compression, at the next-stage ATM exchange 10A, as shown in FIG. 7, the frame signal (see FIG. 7(a)) of the TTC 2 Mbps interface sent from PBX 30A constitutes a continuous signal which is directly converted into cells in the AAL (ATM Adaptation Layer) type 1 before being transferred.
An ATM cell (see FIG. 7(b)) generated by this cell conversion consists of a total of 53 bytes, consisting of a 5-byte "ATM header", in which control information such as its destination is stored, and a 48-byte "information field" containing user data. An ATM cell in which a TTC 2 Mbps interface frame is directly inserted in this information field is transferred from ATM exchange 10A to remote ATM exchange 10B.
In contrast, when transfer is effected by PBX 30A with voice compression, at ATM exchange 10A, as shown in FIG. 8, each of the ch on the TTC 2 Mbps interface frame signal (see FIG. 8(a)) sent from PBX 30A, for example call control ch (see FIG. 8(b)), the 16 kbps compressed voice of voice ch1 (see FIG. 8(c)), the 16 kbps compressed voice of voice ch2, . . . and the 16 kbps compressed voice (see FIG. 8(d)) of voice ch30 are separately converted to ATM cell form and transferred to the remote party.
As described above, when a voice signal was transferred through an ATM exchange without compression, it normally occupied a bandwidth of 2 Mbps irrespective of whether or not a voice call was taking place or the busy settings. But when voice compression is employed, since the voice bandwidth can be set to 1/4 and further such that channels that are not busy can be set beforehand as non-transmitting, a transfer bandwidth much smaller than 2 Mbps is sufficient. However, if voice compression is employed, celling is performed in units of each ch, so, in the case of 16 kbps, the data rate is low, with the result that time is required for the accumulation of data amounting to one cell. Consequently, in order to compile one cell of voice data as shown in FIG. 8(b), (c) and (d), 125 .mu.sec.times.(47 byte.div.1/4)=23.5 msec is required. Including the time required for voice compression processing, this gives 50 msec or more, imposing a very large transfer delay.
Also, conventionally, when for example a call is made between telephone set 50-1 and telephone set 50-5, the telephone call originating from telephone set 50-1 is transferred in the sequence: PBX 30a, ATM exchange 10a, ATM exchange 10b, PBX 30b; on determining that this PBX 30b is a relay exchange, in accordance with the result of this determination, the call is then further transferred through relay route 35 in the order: ATM exchange 10b again, ATM exchange 10c, and PBX 30c, after which it arrives at telephone set 50-5. In this case, all the processing of call connection information is performed between PBXs 30a, 30b, and 30c.
Thus, in the case of relay-exchanging, celling and decelling are executed every time relay-exchanging is performed, irrespective of whether voice compression has been applied or not; thus, as the number of times of relay-exchanging increases, the delays accumulate. Consequently, while this may still be satisfactory in the case where voice compression is not applied, if voice compression is applied, in addition to the large delay which is inherent in celling for each channel as described above, there is superimposed the delays resulting from performing celling every time relay-exchanging takes place; as a result, service delays sometimes reached levels that could not be ignored from the point of view of speech quality.
Thus, with the prior art system described above, when performing transfer of non-compressed voice, frame signals were directly converted to cell form before being transferred between the PBX/ATM exchanges, but, when performing compressed voice transfer, the compressed voice in the aforesaid frame signals was converted to cell form for each channel and into TDM frame units of fixed length before being transferred.
In the conventional system described above, there was therefore the problem that, when voice was to be transferred without compression, it was always necessary to reserve bandwidth corresponding to the frame signal between the PBX/ATM exchanges: this adversely affected transfer efficiency.
Also, in the case where transfer was effected after performing voice compression, although the problem of occupation of bandwidth described above did not occur, because celling was effected for each channel, a considerable waiting time was required to accumulate sufficient compressed voice to fill up a TDM frame of fixed length; thus the transfer rate was inevitably lowered due to this celling delay.
Furthermore, with the conventional system described above, when relay-exchanging was performed through a plurality of ATM exchanges, it was necessary to perform celling and decelling every time such relay-exchanging was executed: in particular, in the case of relay-exchanging to transfer compressed voice, the delay of this celling/decelling was superimposed on the delay involved in celling for each channel as described above, corresponding to the number of times that this processing was performed. As a result, the problem arose that transfer delay sometimes reached levels whose effect on service could not be ignored.
Accordingly, an object of the present invention is to provide a voice signal transmitting method and exchange system whereby, in service between telephone sets with an ATM system having a layout that is capable of coping with relay-exchanging, the bandwidth reservation in non-compressed voice transfer can be eliminated and transfer efficiency improved.
A further object of the present invention is to provide a voice signal transmitting method and exchange system wherein, in service between telephone sets in an ATM system having a layout capable of coping with relay-exchanging, the transfer delay resulting from the celling delay in compressed voice transfer and the celling/decelling delay on relay-exchanging can be greatly reduced, enabling excellent speech quality to be maintained.